Cisco 7942 + SIP provider

Similar Questions

• Cannot set Cisco SX20 - SIP Login and password

  Hello

  I have a problem that I can not set up login and password on the SIP profile 1 config page. Whenever I set up and refresh page, login and password is missing and diagnostics show error "401 Unathorized" which register to SIP Proxy (VCS).

  Software version 6.3.

  No idea how it install without factory reset?

  EDIT: I found that if I turn off the SIP protocol, set username and password, it is there. But as soon as I activate SIP back, username and password is missing.

  Thank you

  Hi Tibor,

  This is due to CSCuj48259. If you go into your system and your configuration mode set to OFF, it will solve the problem.

  This defect has not yet been solved, however she is addressed in a future release. I have

  f you run TC 6.2 or higher, you will have this problem if you have your configuration set to auto mode and auto-discovery of a fine return of provisioning succeeds but supplys a blank user name and password for your system. Also, if it is set to auto and DHCP option 150 is configured on this network, he will attempt to provision with CUCM which would clear these fields.

• Driver of Cisco Telepresence - room LoopBack

  Hello world

  Is it possible to create a meeting room on Cisco Telepresence driver that will return the video to the appellant alone in the room? As the [email protected] / * / Cisco's Sip URI? (A kind of selfview room)

  Thanks for your help.

  It is possible to show only one participant connected to a MCU conference selfview, but not for telepresence Server conferences.  It comes without caring if the driver is involved or not.  The service provided by Cisco closure is obtained with the C40/60 endpoints where the audio and the video plays in a loop directly to the connected call (output HDMI to HDMI-in, output audio RCA audio-in RCA/RCA).

• [SPA3102] SIP recording every hour with the 401 error and directly 12 OK

  Location: INET-ADSL modem in bridge mode-SPA3102.
  Problem: not really, everything seems to work, can dial in and out.
  But...
  Because I'm curious, I have a logserver of installation and checked what happened every hour of registration.

  What I see in the syslog hourly the 3102 made re - enroll by the SIP provider, but first I think 401 Unauthorized een 12 error and measured, I see an OK message.

  Seems weird to me.

  The same thing happens when I compose, firstly a 401 that OK.

  Can someone explain why the first attempt gives an error and how to avoid this?

  Some details of syslog:

  I replaced a text in the syslog:

  x.x.x.x is real My outside IP address.
  yyyyyyyyy = my local phone number including the area code, such as 31201234567, 31 = NL, 20 = codeZone for Amsterdam
  username = username by my SIP provider

  message 1:
  REGISTER SIP:SIP.poivy.com SIP/2.0
  Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-2283ef7b
  From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
  To: + 31yyyyyyyyy
  Call ID: [email protected]

  CSeq: 6104 REGISTRY
  Max-Forwards: 70
  Authorization: Digest username = "username", realm = "sip.poivy.com", nonce = "1663445546", uri = "sip:sip.poivy.com", algorithm = MD5 response = "c8c5b94c384559bb490b59b72be1c674"
  Contact: + 31yyyyyyyyy ; expires = 3600
  User-Agent: Linksys/SPA3102-3.3.6(GW)
  Content-Length: 0
  Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFICATION OPTIONS, see
  Support: x-sipura

  Message 2:
  SIP/2.0 401 Unauthorized
  Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-2283ef7b
  From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
  To: + 31yyyyyyyyy
  Contact: sip:x.x.x.x:5060
  Call ID: [email protected]
  CSeq: 6104 REGISTRY
  Server: (very nice Sip Registrar/Proxy Server)
  Allow: ACK, BYE, CANCEL, INVITE, REGISTER, OPTIONS, INFO, MESSAGE
  WWW-Authenticate: Digest realm = "sip.poivy.com", nonce = "1667015687", algorithm = MD5
  Content-Length: 0

  Message 3:
  REGISTER SIP:SIP.poivy.com SIP/2.0
  Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-4d7052c
  From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
  To: + 31yyyyyyyyy
  Call ID: [email protected]

  CSeq: 6105 REGISTRY
  Max-Forwards: 70
  Authorization: Digest username = "username", realm = "sip.poivy.com", nonce = "1667015687", uri = "sip:sip.poivy.com", algorithm = MD5 response = "46f2176652c0ad8d27f8d3ad1cf72c24"
  Contact: + 31yyyyyyyyy ; expires = 3600
  User-Agent: Linksys/SPA3102-3.3.6(GW)
  Content-Length: 0
  Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFICATION OPTIONS, see
  Support: x-sipura

  Message 4:
  SIP/2.0 200 Ok
  Via: SIP/2.0/UDP x.x.x.x:5060; direction = z9hG4bK-4d7052c
  From: + 31yyyyyyyyy ; tag = c85fff819484d288o0
  To: + 31yyyyyyyyy
  Contact: + 31yyyyyyyyy ; expires = 3600
  Call ID: [email protected]

  CSeq: 6105 REGISTRY
  Server: (very nice Sip Registrar/Proxy Server)
  Allow: ACK, BYE, CANCEL, INVITE, REGISTER, OPTIONS, INFO, MESSAGE
  Content-Length: 0

  @hw: thank you for your tip and your right on the spot!

  Never, ever, thought of this way of "logging" in a system and use delberatley an error response. With your tip, I thought that allows to read the RFC and found RFC 3665. This RFC describes the Protocol SIP basic call flow. And there he was, almost at the beginning of the real story on page 5! The protocol uses a command register which gives a message of 401 error back just to be sure to avoid security problems "man in the middle" (if I understand correctly). He present a challenge and you the answer to a totalizer new order including your answer on the challenge. Which will lead to an OK return the message.

  To resume: nothing weird, it is as expected. My curiosity is frankly satisfied. Another day with what I've learned something. Thank you.

• How can I get a trial version of cisco ACS 5.4

  Hi guys:

  I would get a trial version of GBA 5.4 for educational purposes (certification LAB). I know that it is possible to download the ISO file of www.cisco.com, but when a try to download the file with my cisco CCO get a message asking me "an additional fee required. Do you know how can I get this software?

  PD: I was able to download a trial of this software (file *.lic) license, but I want to install the ACS in a VMWARE server and play with him. I need the ISO file.

  Thank you very much for your help

  Kind regards.

  Martin

  CCNA-CCNP-CCGD

  Certified Engineer

  Cisco limited offer of trial copies of some of its products. Those that are linked from here:

  http://www.Cisco.com/go/nmsevals

  In General, if it is not there, it is not available as a trial version. It is usually not Cisco policy to provide all the software trial for teaching and laboratory use.

  If you are working with a Cisco or a partner account manager, you will get an exception on a case-by-case basis.

• How to replace the call number part on a SIP section if none is present

  Hello

  I have a 2821 router that is connected to a SIP trunk.  This router also has a PRI interface that is connected to a PBX Mitel.  The Mitel used the router has a primary connection to the PSTN.

  The Mitel will send the party call number most of the time, but sometimes it isn't.  All calls that are sent to the SIP provider without a party call number are deleted.  I don't know if can substitute part number calling using the command in clid on the Dáil peer, however is it possible only override the calling party number if I do not get one of the Mitel?  He got a number of appeal left like a DID I want to pass along in the SIP call.

  Thank you.

  You should be able to do it with a voice translation rule that matches an empty ANI.
  30rule translation-rule voice 1 / ^ $\+631689\(...\) / / 01689\1/rule 2 / ^ $/ /555666777/
  translation-profile PSTN-OUTtranslate of the voice calling 30

  First rule to match a real number to a PBX, to adapt according to your needs. The second rule matches a null value and changes to 555666777, who adapt to what you would like to send to your phone company.

  Do not forget to rate helpful responses and identify useful or correct answers.

• Cisco IDS 4.1 probes in HA? monitor package drops?

  Hello

  can someone tell me if Cisco IDS sensors provide high availability or failover capabilities? If so, how and where to fix?

  Is there a form any notification drop package when sensor starts a fall of packages under full load?

  Hello

  IDS sensors do not provide high availability or failover capability.

  Under a high load of the sensor can be configured to alert of hamid question the 993 which States "package dropout rate exceeded the threshold. This threshold is set by default to 5% (Total dropped packets / Total packets received in a time interval). You must enable this GIS as it is disabled by default.

  Hope this helps

  Thank you

  Madhu

• IPSec site to site VPN cisco VPN client routing problem and

  Hello

  I'm really stuck with the configuration of ipsec site to site vpn (hub to spoke, multiple rays) with cisco vpn remote client access to this vpn.

  The problem is with remote access - cisco vpn client access - I can communicate with hub lan - but I need also communication of all lans speaks of the cisco vpn client.

  There are on the shelves, there is no material used cisco - routers DLINK.

  Someone told me that it is possible to use NAT to translate remote access IP-lan-HUB customers and thus allow communication - but I'm unable to set up and operate.

  Can someone help me please?

  Thank you

  Peter

  RAYS - not cisco devices / another provider

  Cisco 1841 HSEC HUB:

  crypto ISAKMP policy 1

  BA 3des

  preshared authentication

  Group 2

  ISAKMP crypto key x xx address no.-xauth

  !

  the group x crypto isakmp client configuration

  x key

  pool vpnclientpool

  ACL 190

  include-local-lan

  !

  86400 seconds, duration of life crypto ipsec security association

  Crypto ipsec transform-set esp-3des esp-sha-hmac 1cisco

  !

  Crypto-map dynamic dynmap 10

  Set transform-set 1cisco

  !

  card crypto ETH0 client authentication list userauthen

  card crypto isakmp authorization list groupauthor ETH0

  client configuration address card crypto ETH0 answer

  ETH0 1 ipsec-isakmp crypto map

  set peer x

  Set transform-set 1cisco

  PFS group2 Set

  match address 180

  card ETH0 10-isakmp ipsec crypto dynamic dynmap

  !

  !

  interface FastEthernet0/1

  Description $ES_WAN$

  card crypto ETH0

  !

  IP local pool vpnclientpool 192.168.200.100 192.168.200.150

  !

  !

  overload of IP nat inside source list LOCAL interface FastEthernet0/1

  !

  IP access-list extended LOCAL

  deny ip 192.168.7.0 0.0.0.255 192.168.1.0 0.0.0.255

  deny ip 192.168.7.0 0.0.0.255 192.168.200.0 0.0.0.255

  IP 192.168.7.0 allow 0.0.0.255 any

  !

  access-list 180 allow ip 192.168.7.0 0.0.0.255 192.168.1.0 0.0.0.255

  access-list 190 allow ip 192.168.7.0 0.0.0.255 192.168.200.0 0.0.0.255

  !

  How the DLINK has been configured for traffic between the site to site VPN subnets? You are able to add multiple remote subnets on DLINK? If you can, then you must add the pool of Client VPN subnet.

  Alternatively, if you cannot add multiple subnet on DLINK router, you can change the pool of Client VPN 192.168.6.0/24, and on the crypto ACL between the site to site VPN, you must edit the 180 existing ACL

  DE:

  access-list 180 allow ip 192.168.7.0 0.0.0.255 192.168.1.0 0.0.0.255

  access-list 180 allow ip 192.168.200.0 0.0.0.255 192.168.1.0 0.0.0.255

  TO:

  access-list 180 allow ip 192.168.6.0 0.0.1.255 192.168.1.0 0.0.0.255

  Also change the ACL 190 split tunnel:

  DE:

  access-list 190 allow ip 192.168.7.0 0.0.0.255 192.168.200.0 0.0.0.255

  access-list 190 allow ip 192.168.1.0 0.0.0.255 192.168.200.0 0.0.0.255

  TO:

  access-list 190 allow ip 192.168.7.0 0.0.0.255 192.168.6.0 0.0.0.255

  access-list 190 allow ip 192.168.1.0 0.0.0.255 192.168.6.0 0.0.0.255

  Finally, replace the remote subnet 192.168.7.0/255.255.255.0 192.168.6.0/255.255.254.0 DLINK.

  Hope that helps.

• SIP URI B2B calls via CUCM and Highway

  Hi all

  Cannot find out how to configure the SIP URI from SX codecs, recorded calls on CUCM.

  I CUCM and IM & P 10.5.1 a pair of X8.5.2 expressway with two areas of course - for B2B calls, and the external jabber.

  I want to make calls of SX-codecs outside via SIP, by using the URI field, for example, [email protected] / * / or [email protected] / * /. How can I configure that?

  I already have the routing model, which is the SIP Trunk for Exp - C, for H323 calls by ip address. Is it possible to make a routing model to call the SIP URI?

  You can see my deployment in this image.

  fghj.PNG

  

  Best wishes

  Igor

  Hi Igor

  These guides will help you resolve your issues:

  http://www.Cisco.com/c/dam/en/us/TD/docs/voice_ip_comm/Expressway/config_guide/x8-5/Cisco-Expressway-SIP-trunk-to-unified-cm-deployment-guide-CUCM-8-9-10-and-x8-5.PDF

  https://supportforums.Cisco.com/ru/video/12362936

  BR Oleksandr

• Client VPN sail to 10.0.0.0 when you are connected to the Cisco VPN

  I have a cisco ASA that provides access remote vpn for users. The address pool is 10.16.135.x. When a user (windows) connects to the VPN, a route to 10.0.0.0 255.0.0.0 is added to the routing table. If the private IP address of the user is in the 10.x.x.x range, the user can still connect to its local network. However, if the user has a remote site that also uses 10.x.x.x, the user is no longer able to connect to the remote site because of the 10.0.0.0 new road is added after the vpn connection is established.

  10.0.0.0 255.0.0.0 on a 10.16.135.217 route 281

  10.0.0.0 255.0.0.0 10.0.0.1 10.16.135.217 100

  10.16.135.217 255.255.255.255 on a 10.16.135.217 route 281

  Small,

  Can you please set the pool like this:

  mask 10.16.1.10 - 10.16.1.254 255.255.255.0 IP local pool Subnet_10

  Test again and I would like to know the results.

  Daniel Moreno

  Please note all messages that will be useful

• Cisco 2611 router and RADIUS

  Greetings. First of all, let me start by saying that I am a fool, I know I am a fool and I apologize for wasting everyone's time. In fact, I do RTFM, RTFMs a lot, and I've yet to find a resolution.

  Secondly, I am setting up a RADIUS server in my test network. I installed Yopougon RADIUS on a Windows 2000 System. I have the following Setup on my Cisco 2611 router:

  With the help of 2297 off 29688 bytes

  !

  ! 17:20:27 PDT configuration was last modified Tuesday, May 20, 2008

  ! NVRAM config update at 17:20:29 PDT Tuesday, May 20, 2008

  !

  version 12.1

  no single-slot-reload-enable service

  horodateurs service debug datetime localtime show-timezone msec

  Log service timestamps datetime localtime show-timezone msec

  encryption password service

  !

  host Tester name

  !

  logging buffered debugging 10000

  AAA new-model

  RADIUS AAA server group RadiusServers

  ACCT-port of the server 172.26.0.2 auth-port 1812 1813

  !

  Group AAA authentication login default local RadiusServers

  AAA authentication login local localauth

  AAA authentication ppp default if necessary to group local RADIUS

  AAA authorization exec default local radius group

  RADIUS AAA authorization network default local group

  AAA accounting delay start

  start-stop radius group AAA accounting exec by default

  start-stop radius group AAA accounting network default

  AAA process 6

  Select the secret xxx

  !

  test username password xxx

  !

  clock timezone PST - 8

  clock summer-time recurring PDT

  IP subnet zero

  no ip domain-lookup

  !

  no ip bootp Server

  !

  interface Loopback0

  the IP 192.168.0.1 255.255.255.0

  !

  interface Ethernet0/0

  Description for the main network

  address IP X.X.X.X 255.255.255.128

  no ip redirection

  no ip unreachable

  no ip proxy-arp

  NAT outside IP

  full-duplex

  No cdp enable

  !

  interface Ethernet0/1

  Description of network internal

  IP 172.26.0.1 255.255.255.0

  no ip redirection

  no ip unreachable

  no ip proxy-arp

  IP nat inside

  load-interval 30

  full-duplex

  No cdp enable

  !

  IP nat pool test X.X.X.X-X.X.X.X netmask 255.255.255.128

  IP nat inside source list 3 pool overload test

  IP nat inside destination list 3 pool test

  IP classless

  IP route 0.0.0.0 0.0.0.0 X.X.X.X

  no ip address of the http server

  !

  radius of the source interface Ethernet0/1 IP

  access-list 3 permit 172.26.0.0 0.0.0.255

  not run cdp

  public RO 15 SNMP-server community

  secret key of acct-port 1812 auth-172.26.0.2 - RADIUS server host port 1813

  RADIUS server retransmit 3

  key secret RADIUS server

  !

  Line con 0

  password xxx

  Synchronous recording

  line to 0

  line vty 0 4

  access-class 10

  1234567890 7 password

  Synchronous recording

  !

  NTP-period clock 17208108

  Server NTP 192.43.244.18

  end

  My RADIUS server is in place and respond to queries, but my router does not seem to be transferring applications to authenticate to it. In fact, when I connect to the router using HyperTerminal, it expires, and I find myself authenticate locally.

  I don't really like if my Cisco equipment authenticates with the RADIUS server, but I have to get set up to authenticate my users so that I can follow their time online. What I missed in my router configuration? Therefore no transfer requests to the RADIUS Server user authentication.

  Thanks for any assistance, you may be able to provide.

  If you explore the authentication Proxy and it works, it could make you forget the PPPoE fast enough.

  If you decide to pursue PPPoE, the following link is probably where you will find most of the information on the configuration of Cisco PPPoE:

  http://www.Cisco.com/en/us/Tech/tk175/tk819/tsd_technology_support_protocol_home.html

  "Providers" of Cisco forums could provide some guidance if PPPoE is achievable with your platform and environment?

• CUCME no calls incoming, outgoing calls okay

  Hello everyone,

  I'm setting up a CUCME with SIP trunk, I can make calls outside, but I can´t receive everything from the outside, it's my second time as a SIP configuration

  I ve use debug command voice dialpeer all to check was happening, but I can´t find the problem.

  This is my config:

  IP server host sip - A.B.C.D

  !

  voip phone service

  list of approved IP addresses

  IPv4 A.B.C.D 255.255.255.252

  !

  translation of the voice-rule 1

  rule 1 / 325277\ (\) / / 1\1 /.

  !

  voice translation-profile IN

  translate 1 called

  !

  Dial-peer voice 1 voip

  Description * incoming SIP trunk call *.

  entrants IN translation-profile

  session protocol sipv2

  session target sip-Server

  incoming called-number.

  codec voice-class 1

  voice-class sip dtmf-relay rtp - nte force

  DTMF-relay rtp - nte

  No vad

  !

  ePhone-dn 1

  number 100

  Description of RECEPTION

  !

  ePhone 2

  address Mac YYYY. BENAMER. CCBC

  ePhone-model 1

  type 7942

  Keep-Conference

  button 1:1

  NOTE: The IP address are hidden, just for safety

  Here is the output from my debug/tests:

  voice translation rule 1 32527700 #test

  Matched with rule 1

  Original number: 32527700 translated number: 100

  Number of origin type: no number translation type: no

  Original number plan: no number plan translated: no

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number = 32527700, called number = 32527700, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 32527700

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Dial String = 32527700, expanded String = 32527700 number = 32527700T

  Timeout = TRUE, incoming = FALSE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = 32527700, saf_enabled = 1 saf_dndb_lookup = 1, dp_result =-1

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Number = 59513212, called number =, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_ANSWER; Number = 59513212

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls = 59513212T

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_ORIGINATE; Number = 59513212

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls = 59513212T

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Result = NO_MATCH(-1) after all rules attempt Match

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeer:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Number = 59513212, called number = 32527700, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_VIA_URI; URI = SIP:A.B.C.D:5060

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls =

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_REQUEST_URI; URI = sip:[email protected]/ * /: 5060; user = phone

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls =

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_TO_URI; URI = sip:[email protected]/ * /; user = phone

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls =

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_FROM_URI; URI = sip:[email protected]/ * /; user = phone

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls =

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_INCOMING_DNIS; Called number = 32527700

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Dial String = 32527700, expanded String = 32527700 number =

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/MatchNextPeer:

  Result = Success (0); Incoming dial-peer = 1 is set in correspondence

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Result = Success (0) after DP_MATCH_INCOMING_DNIS; Incoming dial-peer = 1

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerSPI:[email protected]/ * /.

  Can someone help me?

  Thanks in advance!

  I looked on the other leg of the SIP messages appeal, here's the fault for the where incoming call is being failed because the session timer is too small, has received from the SBC (provider)

  Call ID:

  Call ID: [email protected]/ * /.

  INVITE RECEIEVED SBC - SIP

  * Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

  Received:

  GUEST sip: 32527700 @(WAN): 5060; user = phone SIP/2.0

  Via: SIP/2.0/UDP (SIP_SERVER): 5060; Branch = z9hG4bK776928550196f0d843ca0b092

  Call ID: [email protected]/ * /.

  From:; tag = 6e8b9968-CC-25

  TO:

  CSeq: 1 INVITE

  Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, see

  Max-Forwards: 70

  Supported: 100rel, timer

  User-Agent: Huawei SoftX3000 V300R601

  Session time-out: 300

  Min - SE: 90

  Contact:

  Content-Length: 376

  Content-Type: application/sdp

  v = 0

  o = HuaweiSoftX3000 4507886 4507886 IN IP4 (SIP_SERVER)

  s = call Sip

  c = IN IP4 (SIP_SERVER)

  t = 0 0

  m = audio RTP 11554 / AVP 8 0 18 4 2 98 98 98

  a = rtpmap:8 PCMA/8000

  a = rtpmap:0 PCMU/8000

  a G729/8000 rtpmap:18 =

  a = rtpmap:4 G723/8000

  a = rtpmap:2 G726-32/8000

  a = rtpmap:98 G726-40/8000

  a = rtpmap:98 G726-32/8000

  a = rtpmap:98 G726-24/8000

  a = ptime:20

  a = fmtp:18 annex b = No.

  In response to GUY sends 422

  Envoy:

  SIP/2.0 422 Session Timer too small

  Via: SIP/2.0/UDP (SIP_SERVER): 5060; Branch = z9hG4bK776928550196f0d843ca0b092

  From:; tag = 6e8b9968-CC-25

  Up to:; tag = 4CD1E84-2094

  Date: Wednesday, January 29, 2014 22:53:19 GMT

  Call ID: [email protected]/ * /.

  CSeq: 1 INVITE

  Allow-events: telephone-event

  Min - SE: 1800

  Server: Cisco-SIPGateway/IOS-15.2.4.M

  Content-Length: 0

  According to rfc

  If the Session time-out interval is too low for a proxy (i.e., lower)

  that the value Min - SE that the proxy would argue), the

  Proxy denies the request with a 422 response.  This response

  contains a header field in Min - TO identify the minimum session

  meantime, she is ready to support.  The UAC will try again, this time

  including the header of Min - SE in the query field.  The header field

  contains the largest header field Min - SE that he observed in all 422

  responses received previously.  In this way, the minimum timer meets the

  constraints of all proxies on the way.

  http://www.Cisco.com/en/us/docs/iOS/voice/SIP/configuration/guide/sip_cg-msg_tmr_rspns.html#wp1056968

  Response message 422

  If the value of the Session header expires is too small, the UAS or proxy refuses the call with a response message 422 Session Timer too small . With 422 response, the proxy or the SAMU message includes a header of Min - SE, indicating the value of minimum session, he can accept. UAC can then try again the appeal with a higher value of session timer.

  If a 422-response message is received after a GUEST query, the UAC can again INVITE him.

  There is two way to fix this

  1) asked the SBC (your SIP provider) the value change and the value of standards send the SIP invite session expires

  (2) change the value of the Min - SE on the CME on demand

  Run this Global Config on CME

  voip phone service

  allow sip to sip connection

  SIP

  90 min - to

  BR,

  Nadeem

  Please note all the useful post.

• PAP2T - calling for not having tone on incoming calls

  After several rounds of configuration, almost everything seems to work on my setup using the Cisco Linksys PAP2T-NA VoIP. The only problem that I face now is that anyone calling my number will not get a dial tone, even if my phone rings normally and if I have the phone, the call passes by normally pickup. But not having is not the tone on the side of the appellant sometimes makes them wonder if the call was successful, and then sometimes they will eventually cling.

  My SIP provider is justvoip.com and my DID is didww.com

  Anything else I should look?

  Thank you very much!

  Ricardo.

  The "target phone rings", also known as the "reminder" signal is sent into the goal of management-> caller.

  There are two basic types of message SIP world.

  The first (type 180) is just message - «target sounds» This type of signal wear no special tone. It is the appellant PBX (or phone same visitor) to play the sound signal.

  The second (type 183) message "I'll give you tones of progress to appellant by oneself." This type of signal begins one-way audio channel to the caller, tones of progress, including the ringing of waiting, can be passed trough it.

  The first type is preferred. The reminder signal is generated alongside the appellant - so the known caller signal is generated (also no bandwidth is consumed to complete). The second type is mainly used to transport non-standard signal, oral messages or so.

  To complicate matters, any PBX on the path can consider changing the message type 180 to 183 and vice versa.

  PAP2T responds 180 all the time. According to your description, it seems that someone considered as ask "I'll play the audio signals", but then it plays no signal. Or a person considered cutting the audio (not every operator allow audio to pass through its network beginning) but has not returned 183 message message 180.

  It may be your VoIP provider, it is the upstream provider, the provider carrying the call to PSTN, an operator of transit to the PSTN and same origin operator RTC on call side.

  Nobody will not disclose the path to the appeal to you - it is considered to be trade secret. And even if you know it, you are not eligible to call all operators asking them to do something. You have no business relationship for them, or you can provide details of calls (known for 'previous-in-path' operator only) for them.

  As I said in a previous comment - if the calling user is affected by a problem, then it must report it to their operator. This operator will review the question and ask the jump according to operator assistance if necessary. Same procedure is repeated until an operator at the origin of the problem will be reached.

  Well, you can complain to your carrier, but it is may not be able to help.

• conecting the 7921g wireles phone to a srp527w?

  Hello

  Anyone know if it is possible to integrate a voip wireless cisco 7921g (sccp protoclol) with a small bussiness Srp527w (sip Protocol) gateway, even if if is by conecting another device (such as an access point) between the two? I know skiny is owned by cisco and sip is a protocol open source, so I don't understand why cisco does not implement the skinny Protocol in its Srp527w.

  Thank you

  Hi Javier, the reason is, most of the Cisco small business product line does not support exclusive protocols of Cisco with the exception of the CDP. 7921 phone is considered a "business phone" and is supported by Cisco TAC, not the HWC.

  ESP is considered a SAI that can register for analog phones to a SIP provider while also using local telco access integrated as a DSL termination and maintenance as a router.

  As mentioned previously, you should have no problem to wireless connect the phone to the RPS. But you will have problems using the signalling Protocol.

  Andrew Hickman has perhaps some information better than me. I'll hit up and see if there is a more clear information, better than what I provided.

  -Tom
  Please mark replied messages useful

• Accidentally deleted network card

  I just bought the Cisco Linksys WRT54G2 provided with the Cisco Network Magic Pro. There is a map that shows every time that a new device connects to my router. Then I connected with my Ipod touch.

  Accidentally, I followed my Ipod touch as an intruder and removed the card. Now my ipod touch cannot find the network "Linksys" but "unable to join the network.

  Please help thanks

  Hi, disable the network lock, and then try to connect your iPod again.